similar to: DTMF A-D

Assuming the price of an ADSI screen phone (say, Aastra 390) was the same as an IP screen phone (say, Cisco 7960) and someone was setting up an * server for their 20 employees (each of whom would have either an ADSI or IP phone on their desk), would there be advantages to using the ADSI phones over the IP phones, or vice-versa? For discussion, let's assume that the hardware needed to

Hi, folks. I have a problem using Asterisk 1.2. I create conferences using app_meetme and Zap channels, and for every participant I run the script defined by AGI_BACKGROUND_SCRIPT to be able to listen and react to DTMF tones. As the docs tell me, when using the AGI background script one loses the ability to control the meetme conference via the command line so for muting conference participants I

Hi all, When the Dahdi driver detects DTMF, it seems it's not muting the first 5-15 ms and sometimes the last 2-10 ms of the DTMF tone. This shows up in recorded voicemail greetings -- you hear a very short DTMF '#', or sometimes two blips, at the end of the recording. I have a Mitel SX-200 connected to Asterisk 1.6.0.1 by a couple of Digium cards: a TE420 w/Octasic and pri_net

Is there any way to program a soft key in ADSI to put a caller on hold. Then able to retreive that caller. Example - Softkey Hold Softkey Retreive Call Softkey End Call -gcc

Hey all, I recently noticed that DTMF caller-ID was implemented in CVS, so I requested the service from my telco (the Dutch KPN) and tried to get it going in Asterisk (current CVS), without success so far. This system has 1 X100P, 2 TDM400P's with 4 FXS-modules each and 2 HFC-PCI ISDN-cards (zaphfc-driver) in it. The analog line I'm trying to get caller-ID working on is obviously on the

Ok trying this again... is there anyone using the SPA-3000 with * ???? I am not sure if this is a specific problem to it or not. This is something I really need to fix!!! When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot access (reliably) DTMF menus at the called party, after call completion. Dialing DTMF is fine. I checked by calling myself. Listening to either end on a

I know I heard people looking for inexpensive ADSI phones a while back. Can't vouch for these guys, but this looks like a reasonable deal for what perhaps are new phones. No experience with ADSI, myself, but thought I'd pass it along. These folks also carry the Nortel analog units that do intercom and 2-line, as has been discussed in prior threads. JT ---------------------

Ok, so I figured out my problem with my pt480s. But, now I have a few more. 1. When I dial into the voicemailmain or voicemailmain2 application, the phone and * start talking adsi, but then the phone tells me "programming download canceled, services is full.", but my services list isn't full, only "Asterisk PBX" occupies slot 2, slots 1, 3 and 4 are available. Any ideas?

I have the original version of the IAXY. I had it laying around collecting dust, now Im actually putting it to use. When I call my voicemail extension (8500), Before I get the voice prompts from the voicemail app, I hear tones that sound like the caller id tones that are heard when montoring a phone call. While watching my Asterisk CLI, I see this error at the sound of each tone: Jul 21 23:06:03

Still grasping at straws trying to solve DTMF detection issues with one of my asterisk servers. This particular server is now running Asterisk 1.4.18.1 and Zaptel 1.4.9.2 in runlevel 3 (console only) with 2 X100P cards. I have tried adjusting channel gains, turning call progress and relaxdtmf on and off, switching echo cancelers, just about everything that Google turns up and I can't

Hi, Would anyone know of a way to set the time automatically on an ADSI capable phone from *? The phone in question is a Aastra 480e. While I am at it, does anyone have any helpful docs on the ADSI script programming? I have managed to do basic functions by modifying the asterisk.adsi file using stuff gleaned from the app_adsiprog.c file, but docs would be really helpful at this point. Tia,

I am having difficulties with customer-bound DTMF being very short & clipped off (and basically unusable, as systems on the customer side aren't recognizing the DTMF digits, and I can barely tell that DTMF is there when I listen on a handset). My system set up as follows: PSTN <--> Metaswitch <-SIP-> Asterisk <-SIP or IAX2-> CPE Asterisk is running Asterisk 10.4.0 on a

Hi, I am using a channel bank and zaptel hardware. I have a credit card machine on one of the channels that appears to be dialing "too soon" for asterisk, every complete number recognized by asterisk is missing the first 1-4 numbers. This is a serious problem for me, anyone have any ideas on whats going on? The pstn picks up on the dtmf tones just fine. I was able to get it to

Wondering, since they appear to be plentiful on eBay, whether I could get a Nortel 350 to use to learn my way around ADSI. The vendor claims that these are "generic," and looking through the archives I wonder if that means that they might be unlocked in the sense that the word is meaningful to asterisk. Of course I am green as could be on this topic, so this question may even be a

Hi, I've recently installed Asterisk and have got the majority of it configured (what an excellent piece of software it is, too), but I'm having a couple of problems. The first one is with music on hold! I've downloaded and installed mpg123 as specified: ># whereis mpg123 >mpg123: /usr/local/bin/mpg123 It's the correct version: >#

I am curious. I understand that features can be pushed to an ADSI phone that make navigating your own voicemail easier, and for other internal things. But, does anyone push this data outside of their own phone network? Example: I am at home with my spiffy new ADSI compatible handset and dial up my bank, OneWorldBankingConglomerate. Would they be able to push me their menu? Button 1 for

Hi list! ConfBridge dtmf_passthrough=no doesn't seem to have any effect. DTMF gets transmitted throughout the conference. I've tried Asterisk 10.7.1 from the official RPMs and 10.8.0 compiled from source. I've confirmed that it's disabled via the CLI "confbridge show profile user <profilename>". It's an all-SIP scenario with RFC2833 as the DTMF protocol.

Hi.. I just wondering why DTMF are not recognized by aterisk on incoming calls from my SIP provider ... ANy suggesteions ?` /Mike

I bought an Aastra PT480 from digium, but I wanted to see if I could get some more help with this before Monday. Any help would be appreciated. I have the phone connected to the TDM400P card, and I also have the T100P and the X100P in the same box. My problem is, it appears as if the phone and asterisk can't understand each other. The port the phone is connected to always remains

What kind of channelbank/FXS port are you connecting to? I've seen problems connecting to some of the older versions of the Adtran Total Access 750's. I wouldn't doubt there would be problems on other channelbanks with older firmwares. Of course, no firmware on CAC AB1's I have the AAstra 480, Adtran 750 Channelbank (updated firmware), T100P card, and it worked fine on the